An Effective Storage Method During A Sampling of Speech Signals

음성신호를 표본화할 동안 효율적인 실시간 저장기법

It is necessary for the speech samples to be stored in memory buffer before speech analyzers without a real time processor process them. In this paper, we propose an algorithm that uses the buffer efficiently, when the analog speech signal is converted to the digital samples by the analog to digital converter. In order to implement this method in real time, the buffer is divided into the starting buffer and the remaining buffer. Until a voiced speech is found, the converted samples are sequentially stored in the starting buffer, and then the buffer is shifted. When a voiced speech is found, the next samples are sequentally recorded in the remaining buffer.